2 * This file is part of DisOrder.
3 * Copyright (C) 2007, 2008 Richard Kettlewell
5 * This program is free software: you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License as published by
7 * the Free Software Foundation, either version 3 of the License, or
8 * (at your option) any later version.
10 * This program is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
13 * GNU General Public License for more details.
15 * You should have received a copy of the GNU General Public License
16 * along with this program. If not, see <http://www.gnu.org/licenses/>.
18 /** @file clients/playrtp.c
21 * This player supports Linux (<a href="http://www.alsa-project.org/">ALSA</a>)
23 * href="http://developer.apple.com/audio/coreaudio.html">Core Audio</a>)
24 * systems. There is no support for Microsoft Windows yet, and that will in
25 * fact probably an entirely separate program.
27 * The program runs (at least) three threads. listen_thread() is responsible
28 * for reading RTP packets off the wire and adding them to the linked list @ref
29 * received_packets, assuming they are basically sound. queue_thread() takes
30 * packets off this linked list and adds them to @ref packets (an operation
31 * which might be much slower due to contention for @ref lock).
33 * The main thread is responsible for actually playing audio. In ALSA this
34 * means it waits until ALSA says it's ready for more audio which it then
35 * plays. See @ref clients/playrtp-alsa.c.
37 * In Core Audio the main thread is only responsible for starting and stopping
38 * play: the system does the actual playback in its own private thread, and
39 * calls adioproc() to fetch the audio data. See @ref
40 * clients/playrtp-coreaudio.c.
42 * Sometimes it happens that there is no audio available to play. This may
43 * because the server went away, or a packet was dropped, or the server
44 * deliberately did not send any sound because it encountered a silence.
47 * - it is safe to read uint32_t values without a lock protecting them
53 #include <sys/socket.h>
54 #include <sys/types.h>
55 #include <sys/socket.h>
61 #include <netinet/in.h>
70 #include "configuration.h"
80 #include "inputline.h"
83 #define readahead linux_headers_are_borked
85 /** @brief Obsolete synonym */
86 #ifndef IPV6_JOIN_GROUP
87 # define IPV6_JOIN_GROUP IPV6_ADD_MEMBERSHIP
90 /** @brief RTP socket */
93 /** @brief Log output */
96 /** @brief Output device */
99 /** @brief Minimum low watermark
101 * We'll stop playing if there's only this many samples in the buffer. */
102 unsigned minbuffer = 2 * 44100 / 10; /* 0.2 seconds */
104 /** @brief Buffer high watermark
106 * We'll only start playing when this many samples are available. */
107 static unsigned readahead = 2 * 2 * 44100;
109 /** @brief Maximum buffer size
111 * We'll stop reading from the network if we have this many samples. */
112 static unsigned maxbuffer;
114 /** @brief Received packets
115 * Protected by @ref receive_lock
117 * Received packets are added to this list, and queue_thread() picks them off
118 * it and adds them to @ref packets. Whenever a packet is added to it, @ref
119 * receive_cond is signalled.
121 struct packet *received_packets;
123 /** @brief Tail of @ref received_packets
124 * Protected by @ref receive_lock
126 struct packet **received_tail = &received_packets;
128 /** @brief Lock protecting @ref received_packets
130 * Only listen_thread() and queue_thread() ever hold this lock. It is vital
131 * that queue_thread() not hold it any longer than it strictly has to. */
132 pthread_mutex_t receive_lock = PTHREAD_MUTEX_INITIALIZER;
134 /** @brief Condition variable signalled when @ref received_packets is updated
136 * Used by listen_thread() to notify queue_thread() that it has added another
137 * packet to @ref received_packets. */
138 pthread_cond_t receive_cond = PTHREAD_COND_INITIALIZER;
140 /** @brief Length of @ref received_packets */
143 /** @brief Binary heap of received packets */
144 struct pheap packets;
146 /** @brief Total number of samples available
148 * We make this volatile because we inspect it without a protecting lock,
149 * so the usual pthread_* guarantees aren't available.
151 volatile uint32_t nsamples;
153 /** @brief Timestamp of next packet to play.
155 * This is set to the timestamp of the last packet, plus the number of
156 * samples it contained. Only valid if @ref active is nonzero.
158 uint32_t next_timestamp;
160 /** @brief True if actively playing
162 * This is true when playing and false when just buffering. */
165 /** @brief Lock protecting @ref packets */
166 pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
168 /** @brief Condition variable signalled whenever @ref packets is changed */
169 pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
171 #if DEFAULT_BACKEND == BACKEND_ALSA
172 # define DEFAULT_PLAYRTP_BACKEND playrtp_alsa
173 #elif DEFAULT_BACKEND == BACKEND_OSS
174 # define DEFAULT_PLAYRTP_BACKEND playrtp_oss
175 #elif DEFAULT_BACKEND == BACKEND_COREAUDIO
176 # define DEFAULT_PLAYRTP_BACKEND playrtp_coreaudio
179 /** @brief Backend to play with */
180 static void (*backend)(void) = DEFAULT_PLAYRTP_BACKEND;
182 HEAP_DEFINE(pheap, struct packet *, lt_packet);
184 /** @brief Control socket or NULL */
185 const char *control_socket;
187 /** @brief Buffer for debugging dump
189 * The debug dump is enabled by the @c --dump option. It records the last 20s
190 * of audio to the specified file (which will be about 3.5Mbytes). The file is
191 * written as as ring buffer, so the start point will progress through it.
193 * Use clients/dump2wav to convert this to a WAV file, which can then be loaded
194 * into (e.g.) Audacity for further inspection.
196 * All three backends (ALSA, OSS, Core Audio) now support this option.
198 * The idea is to allow the user a few seconds to react to an audible artefact.
200 int16_t *dump_buffer;
202 /** @brief Current index within debugging dump */
205 /** @brief Size of debugging dump in samples */
206 size_t dump_size = 44100/*Hz*/ * 2/*channels*/ * 20/*seconds*/;
208 static const struct option options[] = {
209 { "help", no_argument, 0, 'h' },
210 { "version", no_argument, 0, 'V' },
211 { "debug", no_argument, 0, 'd' },
212 { "device", required_argument, 0, 'D' },
213 { "min", required_argument, 0, 'm' },
214 { "max", required_argument, 0, 'x' },
215 { "buffer", required_argument, 0, 'b' },
216 { "rcvbuf", required_argument, 0, 'R' },
217 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
218 { "oss", no_argument, 0, 'o' },
220 #if HAVE_ALSA_ASOUNDLIB_H
221 { "alsa", no_argument, 0, 'a' },
223 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
224 { "core-audio", no_argument, 0, 'c' },
226 { "dump", required_argument, 0, 'r' },
227 { "socket", required_argument, 0, 's' },
228 { "config", required_argument, 0, 'C' },
232 /** @brief Control thread
234 * This thread is responsible for accepting control commands from Disobedience
235 * (or other controllers) over an AF_UNIX stream socket with a path specified
236 * by the @c --socket option. The protocol uses simple string commands and
239 * - @c stop will shut the player down
240 * - @c query will send back the reply @c running
241 * - anything else is ignored
243 * Commands and response strings terminated by shutting down the connection or
244 * by a newline. No attempt is made to multiplex multiple clients so it is
245 * important that the command be sent as soon as the connection is made - it is
246 * assumed that both parties to the protocol are entirely cooperating with one
249 static void *control_thread(void attribute((unused)) *arg) {
250 struct sockaddr_un sa;
256 assert(control_socket);
257 unlink(control_socket);
258 memset(&sa, 0, sizeof sa);
259 sa.sun_family = AF_UNIX;
260 strcpy(sa.sun_path, control_socket);
261 sfd = xsocket(PF_UNIX, SOCK_STREAM, 0);
262 if(bind(sfd, (const struct sockaddr *)&sa, sizeof sa) < 0)
263 fatal(errno, "error binding to %s", control_socket);
264 if(listen(sfd, 128) < 0)
265 fatal(errno, "error calling listen on %s", control_socket);
266 info("listening on %s", control_socket);
269 cfd = accept(sfd, (struct sockaddr *)&sa, &salen);
276 fatal(errno, "error calling accept on %s", control_socket);
279 if(!(fp = fdopen(cfd, "r+"))) {
280 error(errno, "error calling fdopen for %s connection", control_socket);
284 if(!inputline(control_socket, fp, &line, '\n')) {
285 if(!strcmp(line, "stop")) {
286 info("stopped via %s", control_socket);
287 exit(0); /* terminate immediately */
289 if(!strcmp(line, "query"))
290 fprintf(fp, "running");
294 error(errno, "error closing %s connection", control_socket);
298 /** @brief Drop the first packet
300 * Assumes that @ref lock is held.
302 static void drop_first_packet(void) {
303 if(pheap_count(&packets)) {
304 struct packet *const p = pheap_remove(&packets);
305 nsamples -= p->nsamples;
306 playrtp_free_packet(p);
307 pthread_cond_broadcast(&cond);
311 /** @brief Background thread adding packets to heap
313 * This just transfers packets from @ref received_packets to @ref packets. It
314 * is important that it holds @ref receive_lock for as little time as possible,
315 * in order to minimize the interval between calls to read() in
318 static void *queue_thread(void attribute((unused)) *arg) {
322 /* Get the next packet */
323 pthread_mutex_lock(&receive_lock);
324 while(!received_packets) {
325 pthread_cond_wait(&receive_cond, &receive_lock);
327 p = received_packets;
328 received_packets = p->next;
329 if(!received_packets)
330 received_tail = &received_packets;
332 pthread_mutex_unlock(&receive_lock);
333 /* Add it to the heap */
334 pthread_mutex_lock(&lock);
335 pheap_insert(&packets, p);
336 nsamples += p->nsamples;
337 pthread_cond_broadcast(&cond);
338 pthread_mutex_unlock(&lock);
342 /** @brief Background thread collecting samples
344 * This function collects samples, perhaps converts them to the target format,
345 * and adds them to the packet list.
347 * It is crucial that the gap between successive calls to read() is as small as
348 * possible: otherwise packets will be dropped.
350 * We use a binary heap to ensure that the unavoidable effort is at worst
351 * logarithmic in the total number of packets - in fact if packets are mostly
352 * received in order then we will largely do constant work per packet since the
353 * newest packet will always be last.
355 * Of more concern is that we must acquire the lock on the heap to add a packet
356 * to it. If this proves a problem in practice then the answer would be
357 * (probably doubly) linked list with new packets added the end and a second
358 * thread which reads packets off the list and adds them to the heap.
360 * We keep memory allocation (mostly) very fast by keeping pre-allocated
361 * packets around; see @ref playrtp_new_packet().
363 static void *listen_thread(void attribute((unused)) *arg) {
364 struct packet *p = 0;
366 struct rtp_header header;
373 p = playrtp_new_packet();
374 iov[0].iov_base = &header;
375 iov[0].iov_len = sizeof header;
376 iov[1].iov_base = p->samples_raw;
377 iov[1].iov_len = sizeof p->samples_raw / sizeof *p->samples_raw;
378 n = readv(rtpfd, iov, 2);
384 fatal(errno, "error reading from socket");
387 /* Ignore too-short packets */
388 if((size_t)n <= sizeof (struct rtp_header)) {
389 info("ignored a short packet");
392 timestamp = htonl(header.timestamp);
393 seq = htons(header.seq);
394 /* Ignore packets in the past */
395 if(active && lt(timestamp, next_timestamp)) {
396 info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32,
397 timestamp, next_timestamp);
400 /* Ignore packets with the extension bit set. */
401 if(header.vpxcc & 0x10)
405 p->timestamp = timestamp;
406 /* Convert to target format */
407 if(header.mpt & 0x80)
409 switch(header.mpt & 0x7F) {
411 p->nsamples = (n - sizeof header) / sizeof(uint16_t);
413 /* TODO support other RFC3551 media types (when the speaker does) */
415 fatal(0, "unsupported RTP payload type %d",
419 fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n",
420 seq, timestamp, p->nsamples, timestamp + p->nsamples);
421 /* Stop reading if we've reached the maximum.
423 * This is rather unsatisfactory: it means that if packets get heavily
424 * out of order then we guarantee dropouts. But for now... */
425 if(nsamples >= maxbuffer) {
426 pthread_mutex_lock(&lock);
427 while(nsamples >= maxbuffer) {
428 pthread_cond_wait(&cond, &lock);
430 pthread_mutex_unlock(&lock);
432 /* Add the packet to the receive queue */
433 pthread_mutex_lock(&receive_lock);
435 received_tail = &p->next;
437 pthread_cond_signal(&receive_cond);
438 pthread_mutex_unlock(&receive_lock);
439 /* We'll need a new packet */
444 /** @brief Wait until the buffer is adequately full
446 * Must be called with @ref lock held.
448 void playrtp_fill_buffer(void) {
451 info("Buffering...");
452 while(nsamples < readahead) {
453 pthread_cond_wait(&cond, &lock);
455 next_timestamp = pheap_first(&packets)->timestamp;
459 /** @brief Find next packet
460 * @return Packet to play or NULL if none found
462 * The return packet is merely guaranteed not to be in the past: it might be
463 * the first packet in the future rather than one that is actually suitable to
466 * Must be called with @ref lock held.
468 struct packet *playrtp_next_packet(void) {
469 while(pheap_count(&packets)) {
470 struct packet *const p = pheap_first(&packets);
471 if(le(p->timestamp + p->nsamples, next_timestamp)) {
472 /* This packet is in the past. Drop it and try another one. */
475 /* This packet is NOT in the past. (It might be in the future
482 /** @brief Play an RTP stream
484 * This is the guts of the program. It is responsible for:
485 * - starting the listening thread
486 * - opening the audio device
487 * - reading ahead to build up a buffer
488 * - arranging for audio to be played
489 * - detecting when the buffer has got too small and re-buffering
491 static void play_rtp(void) {
495 /* We receive and convert audio data in a background thread */
496 if((err = pthread_create(<id, 0, listen_thread, 0)))
497 fatal(err, "pthread_create listen_thread");
498 /* We have a second thread to add received packets to the queue */
499 if((err = pthread_create(<id, 0, queue_thread, 0)))
500 fatal(err, "pthread_create queue_thread");
501 /* The rest of the work is backend-specific */
505 /* display usage message and terminate */
506 static void help(void) {
508 " disorder-playrtp [OPTIONS] ADDRESS [PORT]\n"
510 " --device, -D DEVICE Output device\n"
511 " --min, -m FRAMES Buffer low water mark\n"
512 " --buffer, -b FRAMES Buffer high water mark\n"
513 " --max, -x FRAMES Buffer maximum size\n"
514 " --rcvbuf, -R BYTES Socket receive buffer size\n"
515 " --config, -C PATH Set configuration file\n"
516 #if HAVE_ALSA_ASOUNDLIB_H
517 " --alsa, -a Use ALSA to play audio\n"
519 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
520 " --oss, -o Use OSS to play audio\n"
522 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
523 " --core-audio, -c Use Core Audio to play audio\n"
525 " --help, -h Display usage message\n"
526 " --version, -V Display version number\n"
532 int main(int argc, char **argv) {
534 struct addrinfo *res;
535 struct stringlist sl;
537 int rcvbuf, target_rcvbuf = 131072;
540 struct ipv6_mreq mreq6;
542 char *address, *port;
546 struct sockaddr_in in;
547 struct sockaddr_in6 in6;
549 union any_sockaddr mgroup;
550 const char *dumpfile = 0;
552 static const struct addrinfo prefs = {
553 .ai_flags = AI_PASSIVE,
554 .ai_family = PF_INET,
555 .ai_socktype = SOCK_DGRAM,
556 .ai_protocol = IPPROTO_UDP
560 if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
561 while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:R:M:aocC:r", options, 0)) >= 0) {
564 case 'V': version("disorder-playrtp");
565 case 'd': debugging = 1; break;
566 case 'D': device = optarg; break;
567 case 'm': minbuffer = 2 * atol(optarg); break;
568 case 'b': readahead = 2 * atol(optarg); break;
569 case 'x': maxbuffer = 2 * atol(optarg); break;
570 case 'L': logfp = fopen(optarg, "w"); break;
571 case 'R': target_rcvbuf = atoi(optarg); break;
572 #if HAVE_ALSA_ASOUNDLIB_H
573 case 'a': backend = playrtp_alsa; break;
575 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
576 case 'o': backend = playrtp_oss; break;
578 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
579 case 'c': backend = playrtp_coreaudio; break;
581 case 'C': configfile = optarg; break;
582 case 's': control_socket = optarg; break;
583 case 'r': dumpfile = optarg; break;
584 default: fatal(0, "invalid option");
587 if(config_read(0)) fatal(0, "cannot read configuration");
589 maxbuffer = 4 * readahead;
594 /* Get configuration from server */
595 if(!(c = disorder_new(1))) exit(EXIT_FAILURE);
596 if(disorder_connect(c)) exit(EXIT_FAILURE);
597 if(disorder_rtp_address(c, &address, &port)) exit(EXIT_FAILURE);
599 sl.s = xcalloc(2, sizeof *sl.s);
605 /* Use command-line ADDRESS+PORT or just PORT */
610 fatal(0, "usage: disorder-playrtp [OPTIONS] [[ADDRESS] PORT]");
612 /* Look up address and port */
613 if(!(res = get_address(&sl, &prefs, &sockname)))
615 /* Create the socket */
616 if((rtpfd = socket(res->ai_family,
618 res->ai_protocol)) < 0)
619 fatal(errno, "error creating socket");
620 /* Stash the multicast group address */
621 if((is_multicast = multicast(res->ai_addr))) {
622 memcpy(&mgroup, res->ai_addr, res->ai_addrlen);
623 switch(res->ai_addr->sa_family) {
625 mgroup.in.sin_port = 0;
628 mgroup.in6.sin6_port = 0;
633 switch(res->ai_addr->sa_family) {
635 memset(&((struct sockaddr_in *)res->ai_addr)->sin_addr, 0,
636 sizeof (struct in_addr));
639 memset(&((struct sockaddr_in6 *)res->ai_addr)->sin6_addr, 0,
640 sizeof (struct in6_addr));
643 fatal(0, "unsupported family %d", (int)res->ai_addr->sa_family);
645 if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
646 fatal(errno, "error binding socket to %s", sockname);
648 switch(mgroup.sa.sa_family) {
650 mreq.imr_multiaddr = mgroup.in.sin_addr;
651 mreq.imr_interface.s_addr = 0; /* use primary interface */
652 if(setsockopt(rtpfd, IPPROTO_IP, IP_ADD_MEMBERSHIP,
653 &mreq, sizeof mreq) < 0)
654 fatal(errno, "error calling setsockopt IP_ADD_MEMBERSHIP");
657 mreq6.ipv6mr_multiaddr = mgroup.in6.sin6_addr;
658 memset(&mreq6.ipv6mr_interface, 0, sizeof mreq6.ipv6mr_interface);
659 if(setsockopt(rtpfd, IPPROTO_IPV6, IPV6_JOIN_GROUP,
660 &mreq6, sizeof mreq6) < 0)
661 fatal(errno, "error calling setsockopt IPV6_JOIN_GROUP");
664 fatal(0, "unsupported address family %d", res->ai_family);
666 info("listening on %s multicast group %s",
667 format_sockaddr(res->ai_addr), format_sockaddr(&mgroup.sa));
669 info("listening on %s", format_sockaddr(res->ai_addr));
671 if(getsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, &rcvbuf, &len) < 0)
672 fatal(errno, "error calling getsockopt SO_RCVBUF");
673 if(target_rcvbuf > rcvbuf) {
674 if(setsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF,
675 &target_rcvbuf, sizeof target_rcvbuf) < 0)
676 error(errno, "error calling setsockopt SO_RCVBUF %d",
678 /* We try to carry on anyway */
680 info("changed socket receive buffer from %d to %d",
681 rcvbuf, target_rcvbuf);
683 info("default socket receive buffer %d", rcvbuf);
685 info("WARNING: -L option can impact performance");
689 if((err = pthread_create(&tid, 0, control_thread, 0)))
690 fatal(err, "pthread_create control_thread");
694 unsigned char buffer[65536];
697 if((fd = open(dumpfile, O_RDWR|O_TRUNC|O_CREAT, 0666)) < 0)
698 fatal(errno, "opening %s", dumpfile);
699 /* Fill with 0s to a suitable size */
700 memset(buffer, 0, sizeof buffer);
701 for(written = 0; written < dump_size * sizeof(int16_t);
702 written += sizeof buffer) {
703 if(write(fd, buffer, sizeof buffer) < 0)
704 fatal(errno, "clearing %s", dumpfile);
706 /* Map the buffer into memory for convenience */
707 dump_buffer = mmap(0, dump_size * sizeof(int16_t), PROT_READ|PROT_WRITE,
709 if(dump_buffer == (void *)-1)
710 fatal(errno, "mapping %s", dumpfile);
711 info("dumping to %s", dumpfile);